[asterisk-users] RFC 4662 in asterisk 10.12.1
Hello, Is there an implementation for the RFC 4662 for asterisk 10? I found a patch for asterisk 1.8 but nothing for asterisk 10.12. The RFC: "This document presents an extension to the Session Initiation Protocol (SIP)-Specific Event Notification mechanism for subscribing to a homogeneous list of resources. Instead of sending a SUBSCRIBE for each resource individually, the subscriber can subscribe to an entire list and then receive notifications when the state of any of the resources in the list changes." I have a Panasonic KX-UT133 and this phone use this method to BLF. Anyone worked with this issue? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Movistar sip Mexico
Hi, I have Asterisk 10.12.1. I can not figure out the solution. Thank you for your help. Best Regards On Thu, Nov 21, 2013 at 7:07 PM, Alyed wrote: > Which version of Asterisk are you using? > > According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless > you are using Asterisk 10, there's quite some patching (or buying) you'll > need to be doing. > > Alyed > > > 2013/11/21 Bryant Zimmerman > >> Can you funnel them through a specific inbound dial context. Then force a >> re-invite to g729? >> >> Thanks >> >> Bryant Zimmerman (ZK Tech Inc.) >> 616-855-1030 Ext. 2003 >> >> >> -- >> *From*: "Damian Gonzalez" >> *Sent*: Thursday, November 21, 2013 8:25 AM >> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < >> asterisk-users@lists.digium.com> >> *Subject*: Re: [asterisk-users] Movistar sip Mexico >> >> >> Any posible solution? >> >> >> On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner >> wrote: >> >>> It is possible that Asterisk requires an rtpmap even for static payload >>> types (I'm not sure about this). The INVITE from your provider omits >>> rtpmap for payload type 18 (G729) which is perfectly valid. >>> >>> >>> On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez >>> wrote: >>> >>>> Hello, >>>> >>>> Thanks for the quickly response. I have only G729 in the peer but I >>>> have t38pt_udptl= yes >>>> >>>> If I put t38pt_udptl=no , asterisk reject the call with 488 code. >>>> >>>> The problem is that Movistar send T38 codec in all calls and I need >>>> ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have >>>> only T38 I have to negociate a fax call. >>>> >>>> Thanks. >>>> >>>> >>>> On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: >>>> >>>>> Think you only need to make sure you have in your sip.conf file these >>>>> configs: >>>>> >>>>> [your-device-name] >>>>> . >>>>> . >>>>> disallow=all >>>>> allow=g729 >>>>> . >>>>> . >>>>> >>>>> >>>>> Alyed >>>>> >>>>> 2013/11/20 Damian Gonzalez >>>>> >>>>>> Hello, >>>>>> >>>>>> I have a problem with movistar in Mexico with a sip calls. Movistar >>>>>> send to me T38 and G729 in the INVITE and they say that I have to ignore >>>>>> T38 and use G729 in the voice call. >>>>>> >>>>>> When a fax call is made Movistar send only T38 in the INVITE. >>>>>> >>>>>> Invite example: >>>>>> >>>>>> v=0 >>>>>> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 >>>>>> s=sip call >>>>>> c=IN IP4 192.168.1.2 >>>>>> t=0 0 >>>>>> m=audio 6370 RTP/AVP 18 101 >>>>>> a=fmtp:18 annexb=yes >>>>>> a=rtpmap:101 telephone-event/8000 >>>>>> a=fmtp:101 0-15 >>>>>> a=ptime:20 >>>>>> m=image 6372 udptl t38 >>>>>> a=T38FaxVersion:0 >>>>>> a=T38FaxMaxBuffer:1100 >>>>>> a=T38FaxMaxDatagram:612 >>>>>> a=T38MaxBitRate:14400 >>>>>> a=T38FaxRateManagement:transferredTCF >>>>>> a=T38FaxUdpEC:t38UDPRedundancy >>>>>> >>>>>> How can I ignore T38 and use only G729 for this call?. >>>>>> >>>>>> Thanks for your help. >>>>>> >>>>>> Damian >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> >>>>>> -- >>>>>> _ >>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>>>>http://www.asterisk.org/hello >>>>>> >>>>>> asterisk-users mailing list >>>>>> To UNSUBSCRIBE or update options visit: >>>>>>http://lists.digium.
Re: [asterisk-users] Movistar sip Mexico
Any posible solution? On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner wrote: > It is possible that Asterisk requires an rtpmap even for static payload > types (I'm not sure about this). The INVITE from your provider omits > rtpmap for payload type 18 (G729) which is perfectly valid. > > > On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez wrote: > >> Hello, >> >> Thanks for the quickly response. I have only G729 in the peer but I have >> t38pt_udptl= yes >> >> If I put t38pt_udptl=no , asterisk reject the call with 488 code. >> >> The problem is that Movistar send T38 codec in all calls and I need >> ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have >> only T38 I have to negociate a fax call. >> >> Thanks. >> >> >> On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: >> >>> Think you only need to make sure you have in your sip.conf file these >>> configs: >>> >>> [your-device-name] >>> . >>> . >>> disallow=all >>> allow=g729 >>> . >>> . >>> >>> >>> Alyed >>> >>> 2013/11/20 Damian Gonzalez >>> >>>> Hello, >>>> >>>> I have a problem with movistar in Mexico with a sip calls. Movistar >>>> send to me T38 and G729 in the INVITE and they say that I have to ignore >>>> T38 and use G729 in the voice call. >>>> >>>> When a fax call is made Movistar send only T38 in the INVITE. >>>> >>>> Invite example: >>>> >>>> v=0 >>>> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 >>>> s=sip call >>>> c=IN IP4 192.168.1.2 >>>> t=0 0 >>>> m=audio 6370 RTP/AVP 18 101 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> a=ptime:20 >>>> m=image 6372 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38FaxMaxBuffer:1100 >>>> a=T38FaxMaxDatagram:612 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> >>>> How can I ignore T38 and use only G729 for this call?. >>>> >>>> Thanks for your help. >>>> >>>> Damian >>>> >>>> >>>> -- >>>> >>>> >>>> -- >>>> _ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>>http://www.asterisk.org/hello >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> -- >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Kristian Kielhofner > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Movistar sip Mexico
Hello, Thanks for the quickly response. I have only G729 in the peer but I have t38pt_udptl= yes If I put t38pt_udptl=no , asterisk reject the call with 488 code. The problem is that Movistar send T38 codec in all calls and I need ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38 I have to negociate a fax call. Thanks. On Wed, Nov 20, 2013 at 4:46 PM, Alyed wrote: > Think you only need to make sure you have in your sip.conf file these > configs: > > [your-device-name] > . > . > disallow=all > allow=g729 > . > ..... > > > Alyed > > 2013/11/20 Damian Gonzalez > >> Hello, >> >> I have a problem with movistar in Mexico with a sip calls. Movistar send >> to me T38 and G729 in the INVITE and they say that I have to ignore T38 and >> use G729 in the voice call. >> >> When a fax call is made Movistar send only T38 in the INVITE. >> >> Invite example: >> >> v=0 >> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 >> s=sip call >> c=IN IP4 192.168.1.2 >> t=0 0 >> m=audio 6370 RTP/AVP 18 101 >> a=fmtp:18 annexb=yes >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> m=image 6372 udptl t38 >> a=T38FaxVersion:0 >> a=T38FaxMaxBuffer:1100 >> a=T38FaxMaxDatagram:612 >> a=T38MaxBitRate:14400 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxUdpEC:t38UDPRedundancy >> >> How can I ignore T38 and use only G729 for this call?. >> >> Thanks for your help. >> >> Damian >> >> >> -- >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Movistar sip Mexico
Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 6372 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy How can I ignore T38 and use only G729 for this call?. Thanks for your help. Damian -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
